For the purpose of delivering remote teaching or a live event, the transmission of real-time stream data such as images or sounds on the network requires that a transmission bandwidth of the real-time stream data is changed in conformity to an available bandwidth on the network. That is, in the transmission over the best effort network such as the internet, it is expected that the network is congested by traffics of other users. When broadband real-time stream data (hereinafter referred to as “stream data”) is transmitted while the network is thus congested, packet losses frequently occur so that the images or the sounds may be disturbed or interrupted. For that reason, there is a need to transmit the stream data while changing a bandwidth with which the images or the sounds are encoded by an encoder, after an available bandwidth is estimated over the network.
As a method of estimating the available bandwidth over the network, there has been known a method of estimating the bandwidth with the use of a packet loss rate on the network, and a round trip time (RTT: round trip time) between the transmitting terminal and a receiving terminal.
Also, as a method of transmitting the stream data to a large number of receiving terminals in a point-to-multipoint manner, there has been known a method in which the terminals copy and transfer packets to realize a multipoint communication. For example, as that method, there is an application layer multicast (Application Layer Multicast: ALM). In the ALM, a transmitting terminal transmits a packet to a specified receiving terminal, and the specified receiving terminal copies the packet, and transmits the copied packets to plural other receiving terminals. That is, in the ALM, the specified receiving terminal becomes a parent terminal, and copies and transfers the packets to the plural child terminals to realize the multipoint communication. The entire transfer paths among those terminals are called “distribution tree”. In the ALM, because a unicast packet normally used in the internet is used as the packet transmitted between the respective terminals, there is no need to change a router on the network. For that reason, the ALM is advantageous in that the ALM is available over the existing network.
When the stream data is transmitted to the large number of receiving terminals in the point-to-multipoint manner with the available bandwidth on the network, a plurality of paths exist between the transmitting terminal and the receiving terminals. For that reason, there is a need that the specified receiving terminal is selected in the plurality of paths, the bandwidth of the path between the transmitting terminal and the selected receiving terminal is estimated, and the estimated bandwidth is determined as the available bandwidth. Non-patent document 1 has proposed a method in which a receiving terminal worst in a packet loss status is selected as the specified receiving terminal to determine the available bandwidth in the path between the transmitting terminal and the specified receiving terminal. Specifically, Non-patent document 2 has proposed a method in which the available bandwidth is determined with the use of a TFRC (TCP Friendly Rate Control). The TFRC is a method of estimating the available bandwidth with the use of a loss event rate and a round trip time of the packets. In a system of Non-patent document 1, the receiving terminal whose bandwidth is to be estimated is changed every time one receiving terminal worst in the packet loss status is changed to another.
Patent document 1 has proposed a method in which intervals between report packets to be transmitted to a source are controlled to conduct a rate control using the TFRC under a multimedia streaming environment, and also to prevent a problem that the report packets are concentrated on the source. In general, as a time interval at which the packet loss rate is tallied is lengthened, the following property when the network status is changed is deteriorated. For example, when a tallying time of the packet loss rate is intervals of 5 seconds, a status in which the congestion degree of the network is deteriorated cannot be grasped within 5 seconds.
Also, as a method of complementing the packet loss occurring in transmission of stream data on the best effort network such as the internet, an FEC (Forward Error Correction) has been known. For example, in Non-patent document 3, the source generates a redundant code packet from a plurality of packets with the help of an XOR (Exclusive OR) operation, and transmits the redundant code packet to the receiving terminal over the network. For that reason, even if a packet loss occurs on the network, the lost packet of stream data can be regenerated according to the redundant code packet and the packet of the stream data which have arrived at the receiving terminal.
However, the packet loss that can be recovered by the FEC is limited. For example, one redundant code packet is generated from four packets, and if one of those five packets is lost, the lost packet of the stream data can be regenerated from the redundant code packet and the packets of the stream data, which have arrived at the receiving terminal. However, when two or more packets are lost among the five packets, the lost packets of the stream data cannot be regenerated. That is, when the packet loss occurs at given intervals, the packet loss can be recovered by the FEC if the loss rate is 20% or lower, but cannot be recovered by the FEC if the loss rate exceeds 20%. Likewise, when one redundant code packet is generated from three packets, the lost packet of data can be regenerated by the FEC when the loss rate is 25% or lower, but cannot be regenerated by the FEC when the loss rate exceeds 25%.
In the best effort network, when the packet lost in conducting the transmission of the real-time stream data is complemented with the aid of the redundant code packet, an adaptive FEC that changes how to allocate a redundant code according to the loss rate is used. That is, when the adaptive FEC is applied to the above example, one redundant code packet is generated from four packets if the loss rate is 20% or lower, and generated from three packets if the loss rate is from 20% to 25%. That is, when the adaptive FEC is used, a range of the loss rate which can be covered by a specified redundant code allocation system is defined. In the above example, it is assumed that the packet loss occur at the given intervals. In a status in which a random loss occurs, or in a status in which the loss bursts, the range of the loss rate which can be covered with the specified redundant code allocation system is different.
Also, in the best effort network such as the internet, because the traffic volume on the network is not always held constant, the loss rate of the packets fluctuates. For that reason, even when the loss rate viewed from a long-term viewpoint is not changed, the loss rate viewed from a short-term viewpoint may be increased or decreased.